[Freeswitch-users] Webrtc

[Anthony Minessale]

I would be careful with sipml5.  All of the code related to it is GPLv3
which is the new version of the GPL that goes viral over a socket.  So if
you make a product from this code and sell it as a service, you will be
bound to the GPL license and forced to surrender your platform.

See the Web Services section in
http://neptune.billgatliff.com/GPL3.html

Also the project has changed its goals towards making a converter to change
webrtc based connections into regular SIP for people who do not wish to
implement the webrtc profile.......



On Mon, Jan 28, 2013 at 7:57 AM, soni4m <sanjay.sanjaysoni at gmail.com> wrote:

> Hi Anthony,
> Can you please explain how the overall solution will look like once this
> webrtc support in FS effort completes ?
> Will it be the way asterisk is used in many of these demos (One example
> here
> <
> http://highsecurity.blogspot.in/2012/12/webrtc-and-asterisk-11-using-sipml5.html
> >
> )  In a sense that sipml5 (SIP stack in the browser) is necessarily there
> on
> the browser side. My interest here is to understand the topology  here how
> FS will help connecting the legacy SIP devices with browser and what will
> come into the picture in between.
> Thanks
>
>
>
> --
> View this message in context:
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> Sent from the freeswitch-users mailing list archive at Nabble.com.
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-- 
Anthony Minessale II

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